1. Field of the Invention
The present invention is related to reducing transmission overhead in digital telecommunications systems and networks and more particularly, to reducing human to machine overhead in Voice over Internet Protocol (VoIP) based telecommunications.
2. Background Description
State of the art telecommunication systems are digital and, frequently, use Internet Protocol (IP) based communications. Unlike analog voice channels with a continuous analog signal, an IP communications system segments audio data, encodes and packetizes the segments and transmits the encoded IP packets between network entities in a connectionless transfer. Bearing in mind that the human ear has a range of no more than 20 Hertz (20 Hz)-20 KHz and typical telecommunications channels may be only on the order of hundreds of Hz, audio occupies a very small portion of a typical IP communication. Since the minimum sampling rate for a signal to avoid aliasing is twice the highest signal frequency component, a 500 Hz frequency component produces 1000 samples (e.g. 1 KBytes or, for 8 bit samples, 8000 bits) per second. If a single 1 KByte sample is sent every second, there is at least a one second (1 s) latency at the receiving end that is further extended by any transmission delays. Delays between samples cause gaps in the received audio, as well as adding to the latency. So, using packets that are too large and system delays that cause gaps in the transmission such (e.g., causing packet spacing to not be uniform, causes the receiving end audio to halting, fragmented and/or choppy, i.e., what is commonly discussed with Quality of Service (QoS) issues. Trans-Atlantic TV news reports provide common examples of this.
So, standards have been developed and promulgated for Voice over IP (VoIP) communications to insure that typical IP networks compensate for transmission delays and address QoS issues. These standards select adequately small size for audio segments for encoding as relatively small packets and select transmitting those encoded small packets at a relatively high frequency such that decoding and transmission delays are unnoticeable or, at least, tolerable.
G729 is one such standard audio data compression algorithm for VoIP, wherein raw audio is segmented into 10 millisecond segments and each segment is compressed in an IP packet. RFC 3551 defines a net audio data stream for a G729 code/decode (codec) with an 8-kbit/sec data rate. See, e.g., www.apps.ietf.org/rfc/rfc3551.html#sec4.2. Normally, VoIP devices that use the G729 codec, are configured to default for a payload of 20-Bytes/packet with 50-Packets/sec to achieve this 8-kbit/sec data rate. Id.
Real-time Transport Protocol (RTP) packets, for example, include headers that used by IP networks for identification and routing. So, regardless of packet size, 20 or 1000 Bytes, each packet has a fixed overhead. Since packet headers are in addition to and not part of the audio and each packet, regardless of size, includes the same size header, smaller packets incur higher overhead than larger packets. Small packets and high transmission frequency require more channel bandwidth and packet routing and desegmentation requires higher processing capability, i.e., more Machine Instructions per Second (MIPS). Consequently, VoIP communications require a relatively high level of system resources.
Messaging systems, such as voice mail, are common features in modern telecommunications systems. Typically, unanswered calls are routed to voice mail where the caller is greeted with an announcement and/or a voice recorder facility. Although RFC 3551 allows relaxed transfer characteristics that accept higher packetization delays for non-interactive applications (machine-to-machine or browser-to-browser) such as streaming audio/video, IP radio, lectures (webinars) or for links with severe bandwidth constraints, those relaxed transfer characteristics are set by the originating device, e.g., the source of the stream. Such streams based on any G7xx codec use very large RTP packets and may have very large spooler buffer at the receiving end, that spools, perhaps, a few seconds of the media packets.
However, normal VoIP telephony communications between devices in state of the art VoIP communications systems almost always originate with a human, e.g., someone calling from a VoIP phone. The VoIP phone selects transfer parameters for a voice call, i.e., human-to-human. Thus, these human originated calls consume the same level of resources regardless of whether a call is between humans or with a machine, e.g., voice mail. However, reducing the overall consumption of system resources, would allow one to use lower performance systems to handle the same capacity, or achieve increased system capacity for the same system.
Thus, there is a need for reducing VoIP communications overhead, optimizing packet size in VoIP communications system and for minimizing call resource consumption in VoIP communications, especially for human to machine VoIP communications.